In order to effectively utilize radio wave resources in mobile communication systems, it is required to compress speech signals at a low bit rate. On the other hand, it is expected from the user to improve quality of communication speech and implement communication services with high fidelity. In order to implement this, it is preferable not only to improve quality of speech signals, but also to be capable of encoding signals other than speech, such as audio signals having a wider band with high quality.
For such contradictory demands, an approach of hierarchically incorporating a plurality of coding techniques shows promise. Specifically, a configuration is adopted combining in a layered way a first layer encoding section that encodes an input signal using a low bit rate using a model suitable for a speech signal and a second layer encoding section that encodes a residual signal between the input signal and the first layer decoded signal using a model suitable for common signals including the speech signal. Coding schemes having such a layered structure have scalability (capable of obtaining decoded signals even from partial information of bit streams) in bit streams obtained by an encoding section, and such schemes are therefore referred to as scalable coding. The scalable coding has a feature of being capable of also flexibly supporting communication between networks having different bit rates. This feature is suitable for a future network environment where a variety of networks will be integrated with IP protocol.
As conventional scalable coding, for example, there is scalable coding disclosed in Non-Patent Document 1. This document discloses a method where scalable coding is configured using the technique defined in MPEG-4 (Moving Picture Experts Group phase-4). Specifically, at a first layer (base layer), a speech signal—original signal—is encoded using CELP (Code Excited Linear Prediction), and at a second layer (extension layer), a residual signal is encoded using transform coding such as, for example, ACC (Advanced Audio Coder) and TwinVQ (Transform Domain Weighted Interleave Vector Quantization). Here, the residual signal is a signal obtained by subtracting a signal (first layer decoded signal) which is obtained by decoding the encoded code obtained at the first layer, from the original signal.
Non-patent document 1: “Everything for MPEG-4”, written by Miki Sukeichi, published by Kogyo Chosakai Publishing, Inc., Sep. 30, 1998, pages 126 to 127